Reproducing and recording apparatus, decoding apparatus, recording apparatus, reproducing and recording method, decoding method and recording method

ABSTRACT

A reproducing and recording apparatus, a decoding apparatus, a recording apparatus, a reproducing and recording method, a decoding method and a recording method are provided in which a computation for changing acoustic characteristics of compressed digital data is effected on a scale factor of each divided band in the compressed digital data including spectrum data band-divided into a plurality of bands on a frequency axis and the scale factor of every divided band.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a reproducing and recording apparatus,a decoding apparatus, a recording apparatus, a reproducing and recordingmethod, a decoding method and a recording method in which acousticcharacteristics of compressed digital data are changed by effectingcalculation on normalized information in the compressed digital data.

2. Description of the Related Art

Heretofore, there have been a variety of audio signal high-efficiencycoding methods and apparatus, and a few examples of such audio signalhigh-efficiency coding methods and apparatus will be described below.There is known a method called a transform coding method which is one ofblock frequency-band division systems in which an audio signal of a timeregion is blocked at every unit time, a signal of a time axis of everyblock is transformed into a signal on a frequency axis, i.e.quadrature-transformed and then coded at every band. Also, there isknown a method called an SBC (Sub Band Coding) method which is one ofnon-block frequency band division methods in which an audio signal oftime region is not blocked at every time unit but divided into aplurality of frequency bands thereby coded. Further, there is known ahigh-efficiency coding method which is a combination of theabove-mentioned band division coding method and the transform codingmethod. In that case, after the band is divided by the above-mentionedband division coding system, the signal of every band isquadrature-transformed into a signal of a frequency region by theabove-mentioned transform coding system, and the coding is effected atvery orthogonal-transformed band.

As a band-division filter used in the above-mentioned band divisioncoding system, there is known a filter such as QMF (Quadrature Mirrorfilter). This QMF is described in 1976 E. E. Crochiere Digital codingspeed in subbands Bell Syst. Tech. J. Vol. 55, No. 8. 1976. Also, ICASSP83, BOSTON Polyphase Quadrature filters—A new subband coding techniqueJoseph H. Rothweiler describes equal band width filter dividing methodand apparatus such as PQF (Polyphase Quadrature filter).

Also, as the above-mentioned quadrature transform, there is known aquadrature transform in which an input audio signal is blocked at apredetermined unit time (frame) and the time axis is transformed intothe frequency axis by effecting FFT (Fast Fourier Transform) or DCT(Discrete Cosine Transform) or MDCT (Modified Discrete CosineTransform). The above-mentioned MDCT is described in ICASSP 1098Subband/Transform Coding Using Filter Bank Designs Based on Time DomainAliasing Cancellation J. P. Princen A. B. Bradley Univ. of Surrey RoyalMelbourne Inst. of Tech.

Further, as a frequency dividing width used when eachfrequency-band-divided frequency component is quantized, there is knowna band division considering man's auditory characteristics. That is, ina band width in which the band width is widened in the high band on thefrequency axis called a critical band, an audio signal is divided into aplurality of bands, e.g. 25 bands. When data of every band of this timeis encoded, the encoding is executed by a predetermined bit distributionof every band or adaptive bit distribution of every band. For example,when MDCT coefficient data of every band obtained by the MDCT processingis encoded by the bit distribution, the encoding is executed by theadaptive distribution bit number.

Further, in the case of the encoding at every band, data is normalizedat every band and quantized, thereby effecting a so-called blockfloating processing in which a more efficient encoding can be realized.For example, when the MDCT coefficient data obtained by theabove-mentioned MDCT processing is encoded, data is normalized inresponse to the maximum value of the absolute value of theabove-mentioned MDCT coefficient at every band and quantized, therebymaking it possible to execute the more efficient encoding. In thenormalization, there are in advance determined a plurality of numberscorresponding to size information, and the numbers are used asnormalization information. The size information of thepreviously-determined normalization is numbered at an interval of aconstant magnitude.

As the bit distribution method and apparatus therefor, there have beenheretofore known the following two methods.

In the IEEE Transactions of Acoustics, Speech, and Signal Processing,vol. ASSP-25, No. 4, August 1977, bits are distributed on the basis ofthe magnitude of the signal of every band. Further, in the ICASSP 1980The critical band coder—digital encoding of the perceptual requirementsof the auditory system M. A. Kransner MIT, there is described a methodin which a signal-to-noise ratio necessary for every band is obtained byusing an auditory masking and bits are distributed in a fixed fashion.

A signal high-efficiency coded by the above-mentioned method is decodedby the method which follows. Initially, the high-efficiency coded signalis calculated as MDCT coefficient data by using bit distributioninformation of every band, normalization information or the like. TheMDCT coefficient data is transformed into data of time region byso-called IMDCT. When data is band-divided by the band-dividing filterupon encoding, data are further synthesized by using a band-synthesizingfilter. By the above-mentioned operation, data of the original timeregion is decoded.

With respect to the signal of the time region which results fromdecoding the high-efficiency coded signal, let it be considered that themagnitude of the amplitude, i.e. reproduction level is adjusted and thata filter processing which is the level adjustment of every band isexecuted. When the reproduction level is adjusted, such adjustment isrealized by effecting multiplication, addition or subtraction of aconstant amount of the signal component of the time region which is notyet encoded fundamentally or the signal component which is decoded tothe time region. Further, when the filter processing is executed, suchfilter processing is realized by a so-called convolutional computationor a combination of delay circuits and multipliers. In both cases, thereare required a plurality of multipliers, adders, delay circuits and thelike so that the processing process increases.

Also, there is considered a method in which the reproduction level isadjusted by MDCT coefficient data of the MDCT frequency region and thefilter is realized by further adjusting the level. With respect to thismethod, there are required multipliers or adders or multiplication usingthe subtracter or addition or subtraction of the number corresponding tothe number of the MDCT coefficient data so that the processing processincreases.

Further, a similar problem arises when the high-efficiency coded signalis recorded on a certain recording medium and the signal of the timeregion in which the recorded signal is decoded is re-recorded in such amanner that information is changed such that the magnitude of theamplitude, i.e. reproduction level is changed or when information isre-recorded under the condition that information is changed in the formof being processed by the so-called filter effect. In particular, whenthe reproduction level is adjusted in the time region and the adjustedresult is re-recorded on the recording medium, the IMDCT and the MDCTshould be executed so that a quality is deteriorated by computationerror or the like.

A similar problem arises when a filter processing is realized by thetransform to the analog region.

When an analog audio signal is processed by filter processing such as alow-pass filter, a buzz-boost filter, a bandpass filter, a high-passfilter or the like, so-called effect processing, there has heretoforebeen required a special processing IC.

Also, in order to effect the filter processing on a part of audiosignal, after a high-efficiency coded digital audio signal is expandedand a part of the expanded audio signal is processed by a filterprocessing, a resultant audio signal cannot be high-efficiency coded.

SUMMARY OF THE INVENTION

In view of the aforesaid aspect, it is a first object of the presentinvention to provide a reproducing and recording apparatus, a decodingapparatus, a recording apparatus, a reproducing and recording method, adecoding method and a recording method in which an adjustment ofreproduction level of a signal of a time region in which ahigh-efficiency coded signal is decoded can be realized by smallerprocessing process.

It is a second object of the present invention to provide a reproducingand recording apparatus, a decoding apparatus, a recording apparatus, areproducing and recording method, a decoding method and a recordingmethod in which a signal of a time region in which a high-efficiencycoded signal is recorded on a certain recording medium and the recordedsignal is decoded can be re-recorded by smaller processing process whilethe reproduction level is changed and a quality can be prevented frombeing deteriorated when a computation such as IMDCT and MDCT isexecuted.

It is a third object of the present invention to provide a reproducingand recording apparatus, a decoding apparatus, a recording apparatus, areproducing and recording method, a decoding method and a recordingmethod in which a filter processing on a signal of a time region inwhich a high-efficiency coded signal is decoded can be realized bysmaller processing process and simple arrangement and in which a filterprocessing of an arbitrary portion with respect to the time regionsignal becomes possible.

It is a fourth object of the present invention to provide a reproducingand recording apparatus, a decoding apparatus, a recording apparatus, areproducing and recording method, a decoding method and a recordingmethod in which a signal of a time region in which a high-efficiencycoded signal is recorded on a certain recording medium and the recordedsignal is decoded can be re-recorded by smaller processing process andsimple arrangement while information is changed in the form ofinformation with a filter effect achieved thereon, a filter processingof an arbitrary portion with respect to the time region signal becomespossible and in which a quality can be prevented from being deterioratedwhen a computation such as IMDCT and MDCT is executed.

According to an aspect of the present invention, there is provided areproducing and recording apparatus which is comprised of data readmeans for reading compressed digital data including spectrum data whoseband is divided into a plurality of bands on a frequency axis and ascale factor of every divided band from a recording medium, computationmeans for receiving compressed digital data including the band-dividedspectrum data and the scale factor of every divided band from the dataread means and effecting a computation for changing acousticcharacteristics of the compressed digital data, and recording means forre-recording the compressed digital data whose acoustic characteristicsare changed when the computation means computes the scale factor of theevery band on the recording medium.

According to another aspect of the present invention, there is provideda decoding apparatus which is comprised of computation means forreceiving compressed digital data including spectrum data which isband-divided into a plurality of bands on a frequency axis and a scalefactor of every divided band and effecting a predetermined computationon the scale factor of the every divided band, normalization means fornormalizing the band-divided spectrum data contained in the compresseddigital data based on the scale factor of the every divided bandcomputed by the computation means, IMDCT means for obtainingband-divided digital data on a time axis by processing the band-dividedspectrum data normalized by the normalization means in an IMDCT fashion,and band-synthesizing means for band-synthesizing the digital data onthe time axis band-divided by the IMDCT means.

According to other aspect of the present invention, there is provided arecording apparatus which is comprised of MDCT means for processing aninputted digital signal on a time axis in a MDCT fashion to providespectrum data on a frequency axis, scale factor calculating means forcalculating a scale factor of every divided band for normalization byband-dividing the spectrum data on the frequency axis, data compressingmeans for providing compressed data including a scale factor of everydivided band and spectrum data by compressing the spectrum data on thefrequency axis calculated by the scale factor calculating means,computation means for receiving compressed digital data including thescale factor of the every divided band and spectrum data from the datacompressing means and effecting a computation for changing acousticcharacteristics of the compressed digital data on the scale factor ofthe every divided band, and recording means for recording the compresseddigital data in which acoustic characteristics are changed when thecomputation means computes the scale factor of the every band on arecording medium.

According to a further aspect of the present invention, there isprovided a reproducing and recording method which comprises the steps ofreading compressed digital data including spectrum data on a frequencyaxis band-divided and a scale factor of every divided band from arecording medium, effecting a computation for changing acousticcharacteristics of the compressed digital data on the scale factor ofthe every divided band in compressed digital data including the read outspectrum data on the frequency axis band-divided and the scale factor ofevery divided band, and re-recording the compressed digital data whoseacoustic characteristics are changed on the recording medium bycomputing the scale factor of the every band.

According to yet a further aspect of the present invention, there isprovided a decoding method which comprises the steps of effecting apredetermined computation on the scale factor of every divided band incompressed digital data including spectrum data on a frequency axisband-divided into a plurality of bands and the scale factor of everydivided band, normalizing the band-divided spectrum data in thecompressed digital data based on the computed scale factor of the everydivided band, providing digital data on a time axis band-divided byprocessing the normalized band-divided spectrum data in an IMDCTfashion, and band-synthesizing digital data on the time axisband-divided.

In accordance with still a further aspect of the present invention,there is provided a recording method which comprises the steps oftransforming an inputted digital signal on a time axis into spectrumdata on a frequency axis in an MDCT fashion, calculating a scale factorof every divided band for normalization by band-dividing spectrum dataon the frequency axis into a plurality of bands, providing compresseddigital data including a scale factor of every divided band and spectrumdata by compressing the band-divided spectrum data on the frequency axisin response to the calculated scale factor of every divided band,receiving compressed digital data including the scale factor of theevery divided band and spectrum data and effecting a computation forchanging acoustic characteristics of the compressed digital data on thescale factor of the every divided band, and recording the compresseddigital data whose acoustic characteristics are changed on a recordingmedium by computing the scale factor of the every band.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 is a block diagram showing a high-efficiency coding encoder foruse in bit-rate-coding as a specific example according to the presentinvention;

FIG. 2A is a diagram showing a long mode which is an example of astructure of a quadrature transform block used in bit compression;

FIG. 2B is a diagram showing a short mode which is an example of astructure of a quadrature transform block used in bit compression;

FIG. 2C is a diagram showing a middle mode A which is an example of astructure of a quadrature transform block used in bit compression;

FIG. 2D is a diagram showing a middle mode B which is an example of astructure of a quadrature transform block used in bit compression;

FIG. 3 is a block diagram showing an example of a bit assignmentcalculating circuit;

FIG. 4 is a diagram showing a spectrum of a band divided inconsideration of each critical band and a block floating;

FIG. 5 is a diagram showing a masking spectrum

FIG. 6 is a diagram showing results obtained when the minimum audiblecurve and the masking spectrum are synthesized;

FIG. 7 is a diagram showing the manner in which data is coded;

FIG. 8 is a diagram showing details of first byte data in FIG. 7;

FIG. 9 is a block diagram showing a specific example of ahigh-efficiency compression codes signal decoder;

FIG. 10 is a diagram used to explain a low-pass filter;

FIG. 11 is a diagram used to explain a low-pass filter;

FIG. 12A is a block diagram showing a specific example of a recordingapparatus according to the present invention;

FIG. 12B is a block diagram showing an example in which the levels areanalyzed in the specific example of the recording apparatus according tothe present invention;

FIG. 13 is a block diagram showing a specific example of a reproducingapparatus according to the present invention;

FIG. 14 is a block diagram showing a specific example of a transmissionapparatus according to the present invention;

FIG. 15 is a block diagram showing a specific example of a receptionapparatus according to the present invention;

FIG. 16 is a block diagram showing a specific example of a reproducingand recording apparatus according to the present invention;

FIG. 17 is a flowchart to which reference will be made in explaining aspecific example of a reproducing and recording method according to thepresent invention;

FIG. 18 is a flowchart to which reference will be made in explaining aspecific example of a decoding method according to the presentinvention; and

FIG. 19 is a flowchart to which reference will be made in explaining aspecific example of a recording method according to the presentinvention.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

Preferred embodiments of the present invention will hereinafter bedescribed with reference to the drawings.

In the preferred embodiment of the present invention, an input digitalsignal such as an audio PCM (Pulse Code Modulation) signal ishigh-efficiency-coded by technologies such as SBC (Sub Band Coding), ATC(Adaptive Transform Coding) and adaptive bit assignment. Thesetechnologies will be described with reference to FIG. 1 and thefollowing sheets of drawings.

In a specific example of a high-efficiency coding apparatus shown inFIG. 1, an input digital signal is divided into a plurality of frequencybands and spectrum data of obtained frequency axis obtained byquadrature-transforming each frequency band is encoded by adaptivelyassigning bits at every critical band, which will be described later on,in the low-frequency band and at every band which results from furtherdividing the critical band in the middle and high frequency band inconsideration of a block coding effectiveness. In general, theabove-mentioned respective blocks independently become blocks whichgenerate a quantization noise. Further, in the preferred embodiments ofthe present invention, prior to the quadrature-transform, the blocksize, i.e. block length is adaptively changed in response to the inputsignal.

Specifically, as shown in FIG. 1, when a sampling frequency is 44.1 kHz,for example, an audio PCM signal having a frequency of 0 to 22 kHz issupplied to an input terminal 100. This input signal is divided into 0to 11 kHz band and 11 kHz to 22 kHz band by a band-dividing filter 101such as a QMF (Quadrature Mirror Filter). The signal having the bandranging from 0 to 11 kHz is divided by a band-dividing filter 102 suchas QMF filter into 0 to 5.5 kHz band and 5.5 kHz to 11 kHz band.Incidentally, 5.5 kHz, 11 kHz, 22 kHz are obtained by omitting twodecimal place for simplicity. This is also true in the description whichfollows.

The signal of 11 kHz to 22 kHz band from the band-dividing filter 101 issupplied to an MDCT (Modified Discrete Cosine Transform) circuit 103which is an example of the quadrature transform circuit. The signalhaving 5.5 kHz to 11 kHz band from the band-dividing filter 102 issupplied to an MDCT circuit 104, and the 0 to 5.5 kHz band signal fromthe band-dividing filter 102 is supplied to an MDCT circuit 102, inwhich they are MDCT-processed. The MDCT circuits 103, 104, 105 executethe MDCT processing based on block sizes determined by block determiningcircuits 109, 110, 11 provided at everybands. The block size is referredto as a block length, and refers to a width on each time axis dividedwhen the time axis is divided in the quadrature transform.

FIGS. 2A, 2B, 2C, 2D show specific examples of standard input signals ofblock of every band supplied to the respective MDCT circuits 103, 104,105. In the above-mentioned specific examples, three divided filteroutput signals have independently a plurality of quadrature transformblock sizes of every band and their time resolutions are switched basedon a time characteristic of time and a frequency distribution or thelike. In the case of a signal having no sudden large level fluctuation,as shown by the long mode in FIG. 2A, the quadrature transform blocksize is as large as 11.6 ms. In the case of a signal having a suddenlarge level fluctuation, the quadrature transform block size is furtherdivided by two or four. As shown by the short mode in FIG. 2B, when thequadrature transform is all divided by four and the block size is 2.9 msor as shown by the middle mode A in FIG. 2C or as shown by the middlemode B in FIG. 2D, when a part thereof is divided by two and the blocksize is 5.8 mS or when a part thereof is divided by four and the timeresolution is 2.9 mS, it becomes possible to cope with a complex in putsignal in actual practice. In the division of the quadrature transformblock size, if the quadrature transform block size is further divided aslong as the scale of the processing apparatus can be permitted, then thedivision of the quadrature transform block size becomes more effective.The block size is determined by block size determining circuits 109,110, 111 shown in FIG. 1, supplied to the MDCT circuits 103, 104, 105and a bit assignment calculating circuit 118 and outputted from outputterminals 113, 115, 117 as block size information of the correspondingblocks.

Referring back to FIG. 1, in spectrum data on frequency axis obtained bythe MDCT processing in the MDCT circuits 103, 104, 105 or MDCTcoefficient data which are signal components within two-dimensionalblock concerning time and frequency, the low band components aresupplied at every critical band to adaptive bit assignment encodingcircuits 106, 107, 108 and a bit assignment calculating circuit 118 andmiddle band components are further divided in critical band widthconsidering an effectiveness of block floating to the adaptive bitassignment encoding circuits 106, 107, 108 and the bit assignmentcalculating circuit 118. The above-mentioned critical band is afrequency band divided considering man's auditory characteristics and aband which has a noise generated when a pure sound is masked by a narrowband band noise of the same intensity close to the frequency of the puresound. The band width of the critical band is increased in the highfrequency band, and the whole frequency band of 0 to 22 kHz is dividedinto 25 critical bands, for example. The bit assignment calculatingcircuit 118 in FIG. 1 calculates the masking amount of every dividedband considering the effectiveness of the critical band and the blockfloating, energy of every divided band or peak value or the likeconsidering a so-called masking effect or the like based on theabove-mentioned block size information, spectrum data or MDCTcoefficient data. Then, the bit assignment calculating circuit 118calculates the assignment bit number, i.e. bit distribution amount atevery divided band based on calculated results, and transmits the bitdistribution amount to the adaptive bit assignment encoding circuits106, 107, 108 shown in FIG. 1. These adaptive bit assignment encodingcircuits 106, 107, 108 effect re-quantization in which spectrum data orMDCT coefficient data is quantized by normalization in response to thebit number assigned at every divided band considering the block sizeinformation, the critical band and the effectiveness of block floating.The data thus encoded are outputted through output terminals 112, 114,116 shown in FIG. 1. For convenience sake of the following description,each divided band considering the critical band and the effectiveness ofthe block floating is referred to as a unit block.

A specific method of bit assignment executed in the bit assignmentcalculating circuit 118 shown in FIG. 1 will be described next. FIG.3 isa block diagram showing a specific example of the bit assignmentcalculating circuit 118 shown in FIG. 1.

As shown in FIG. 3, the spectrum data on the frequency axis from theMDCT circuits 103, 104, 105 shown in FIG. 1 and the block sizeinformation from the block determining circuits 109, 110, 111 shown inFIG. 1 are supplied to an input terminal 301. Hereinafter, as shown inFIG. 3, the bit assignment calculating circuit 118 executes theprocessing by using the constants, the weighting functions or the likeadaptive to the above-mentioned block size information. As shown in FIG.3, the spectrum data on the frequency axis or the MDCT coefficientinputted from the input terminal 301 is supplied to a band energycalculating circuit 302 which obtains energy of every unit block bycalculating a total sum of amplitude values within the unit block, forexample. It is frequently observed that instead of the energy of everyblock, a peak value, a mean value or amplitude values or the like may beused. As the output from this energy calculating circuit 302, FIG. 4shows a spectrum of a sum total value of respective bands, for example,as SB. However, in FIG. 4, for simplicity, the divided numbers of theunit block are expressed by 12 blocks comprising B1 to B12. Also, theenergy calculating circuit 302 determines normalized data showing theblock floating state of the unit block, i.e. scale factor value whichthe band-compression parameter. Specifically, several positive valuesare prepared as nominated values of the scale factor values in advance.Of these positive values, i.e. of values larger than the maximum valueof the absolute value of the spectrum data or MDCT coefficient withinthe unit block, the minimum value is set to the scale factor value ofthe unit block. The scale factor value is numbered by using several bitsin the form corresponding to the practical value. The above-mentionednumber is stored in a ROM (read-only memory) or the like. The scalefactor values with numbers corresponding to the actual values aredefined to have values at the interval of 2 dB in the above-mentionedsequential order. In the scale factor values determined by theabove-mentioned method in a certain unit block, the above-mentionednumber corresponding to the determined value is used as sub-informationindicating the scale factor of the unit block.

Then, to consider the influence generated in the so-called masking ofthe spectrum SB obtained by the above-mentioned energy calculatingcircuit 302, there is executed a convolution processing in which apredetermined weighting function is multiplied with and added to thespectrum SB. To this end, the output from the energy calculating circuit302 of every band, i.e. respective values of the spectrum SB aresupplied to a convolution filter circuit 303. The convolution filtercircuit 303 comprises a plurality of delay elements for sequentiallydelaying inputted data, a plurality of multipliers for multiplying theoutputs from these delay elements with weighting coefficient, which isthe filter coefficient, and a sum total adder for calculating a sumtotal of the outputs from the respective multipliers. By thisconvolution processing, there is obtained a sum total of the portionsshown by dotted lines in FIG. 4.

The output from the convolution filter circuit 303 is supplied to asubtracter 304. The subtracter 304 calculates a level α corresponding toan allowable noise level, which will be described later on, in theconvoluted region. The level a corresponding to the above-mentionedallowable noise level, i.e. permissible noise level is changed to theallowable noise level of every band of the critical band by effectingthe inverse convolution processing as will be described later on. Theabove-mentioned subtracter 304 is supplied with an admissible functionfor obtaining the above-mentioned level α, i.e. function for expressingthe masking level. The above-mentioned level α can be varied byincreasing or decreasing the numerical value in the above admissiblefunction. The admissible function is supplied from a (n-ai) functiongenerating circuit 305 which will be described next.

That is, the above-mentioned level α is obtained by the followingequation (1) if i is the number sequentially supplied from the low bandof the band of the critical band.

α=S−(n-ai)  (1)

In the equation (1), n and a are constants and a >0. S is the intensityof the convoluted spectrum, and (n-ai) in the equation (1) becomes theadmissible function. By way of example, n=38 and a=1 can be used.

The level a is obtained as described above. The thus obtained level α istransmitted to a dividing circuit 306. The dividing circuit 306 is usedto inverse-convolute the level α in the above-mentioned convolutedregion. When the inverse convolution processing is executed by thedividing circuit 306, an admissible noise spectrum is obtained from theabove-mentioned level α. The admissible noise spectrum is the maskingspectrum. To be more precisely, although the inverse convolutionprocessing requires a complex computation, in the specific example ofthe present invention, the inverse convolution processing is executed byusing the simplified dividing circuit 306.

Next, the above-mentioned masking spectrum is transmitted through asynthesizing circuit 308 to a subtracting circuit 309. The output fromthe energy detecting circuit 302 of every band, i.e. the aforementionedspectrum SB is supplied through a delay circuit 310 to the subtractingcircuit 309. Accordingly, the subtracting circuit 309 subtracts themasking spectrum and the spectrum SB so that, as shown in FIG. 5, in thespectrum SB, the level lower than the level of the masking spectrum MSis masked.

When the above-mentioned synthesizing circuit 308 executes thesynthesis, data indicating a so-called minimum audible curve which isman's auditory characteristic, shown in FIG. 6, supplied from theminimum audible curve generating circuit 307 and the above-mentionedmasking spectrum MS can be synthesized. In this minimum audible curve,if the noise absolute level is lower than this minimum audible curve,then such noise cannot be heard. Although this minimum audible curvebecomes different depending on the difference of reproduction volumeupon reproduction, for example, even when the coding is the same, in thereal digital system, the minimum audible curve is not fluctuated so muchin a music software such as commercially-available CD from a dynamicrange of 16 bits. Therefore, if the quantization noise of the frequencyband close to 4 kHz which is most audible cannot be heard, then it isconsidered that the quantization noise less than the level of thisminimum audible curve cannot be heard in other frequency bands.Accordingly, if a noise near 4 kHz of the word length of the system, forexample, cannot be heard and an admissible noise level, i.e. admissiblequantization coefficient is obtained by synthesizing this minimumaudible curve RC and the masking spectrum MS, the admissible noise levelof this case can be reached to the portion shown by hatch in FIG. 6.Incidentally, in the specific example, the level of 4 kHz of theabove-mentioned minimum audible curve is made coincident with the lowestlevel equivalent to 20 bits, for example. Also, FIG. 6 show a signalspectrum SS at the same time.

After the subtracting processing executed by the subtracting circuit309, the admissible noise level in the output from the subtractingcircuit 309 is corrected based on information of equal loudness curve,for example. Here, the equal loudness curve is the characteristic curveconcerning man's auditory characteristics and is obtained by connectingcurves of sound pressures of sounds at the respective frequencies whoseloudness is the same as that of the pure sound of 1 kHz, for example.The equal loudness curve is also referred to as an equal sensitivitycurve of loudness. Also, this equal loudness curve draws a curvesubstantially the same as that of the minimum audible curve RC shown inFIG. 6. In this equal loudness curve, at the frequency near 4 kHz, ifthe sound pressure is lower than that of 1 kHz by 8 to 10 dB, it can beheard with the same loudness as that of 1 kHz. Conversely, at thefrequency near 50 Hz, if the sound pressure is larger than the of 1 kHzby about 15 dB, it can be heard with the same loudness as that of 1 kHz.Therefore, it is to be appreciated that the admissible coefficient,which the noise exceeding the level of the above-mentioned minimumaudible curve, i.e. admissible noise level may have the frequencycharacteristic obtained by the curve corresponding to the equal loudnesscurve. As described above, to correct the above-mentioned admissiblenoise level in consideration of the equal loudness curve is matched withman's auditory characteristic. Owing to a series of processing executedso far, the admissible noise correction circuit 311 calculated theassignment bits relative each unit block based on various parameter suchas the masking and the auditory characteristic.

Data outputted from this admissible noise correction circuit 311 isoutputted from an output terminal 312 as an output of the bit assignmentcalculation circuit 118 shown in FIG. 1.

That is, in the bit assignment calculation circuit 118 shown in FIG. 1,the system shown in FIG. 3 outputs data in which MDCT output spectrum isprocessed by sub-information as main information and outputs scalefactor indicating state of block floating and word length indicatingword length as sub-information. Based on the above-mentioned data, thescale factor and the word length, the adaptive bit assignment encodingcircuits 106, 107, 108 execute the re-quantization in actual practice,and encode data in accordance with the encoding format.

The normalization information adjustment circuit 119 will be described.As described above, with respect to the scale factor value which isnormalization data, there are prepared in advance several positivevalues as nominated scale factor values. Of these positive values, theminimum value in the values greater than the maximum value of theabsolute value of the spectrum data or MDCT coefficient within the unitblock is set to the scale factor value of the corresponding unit block.The above-mentioned scale factor value is numbered by using several bitin the form corresponding to the actual values. The above-mentionednumber is used as sub-information indicating the scale factor of thecorresponding unit block. The scale factor values corresponding to thenumbered actual values are defined so as to have values at an intervalof 2 dB in the sequential order of the above-mentioned numbers.Accordingly, by changing the above-mentioned number which issub-information indicating the scale factor, it is possible to adjustthe level of the corresponding unit block by 2 dB each. Thenormalization information adjustment circuit 119 is the circuit whichinstructs and outputs numerical values for executing this leveladjustment at every two-dimensional blocks. Moreover, the adders 120,121, 122 are adders for adding the numerical values from thenormalization information adjustment circuit 119 to the above-mentionednumbers which are sub-information indicating the scale factor of theunit block. When the numerical value outputted from the normalizationinformation adjustment circuit 119 is negative, the adders 120, 121, 122act as subtracters.

That is, the level adjustment of 2 dB each becomes possible by addingand subtracting all the same values of the normalization informationadjustment circuit 119 to and from the normalization information of allunit blocks. Further, numerical values are outputted from thenormalization information adjustment circuit 119 at everytwo-dimensional block and added to and subtracted from the block whichis to be adjusted in level, the level adjustment of 2 dB each becomespossible independently at every two-dimensional blocks.

The level adjustment is executed independently at every two-dimensionalblocks as described above, thereby making it possible to realize afilter effect.

Incidentally, the added and subtracted results are limited such thatthey are confined within the range of the scale factor determined by theformat.

Then, the data encoding format in which the encoding is executed inactual practice will be described with reference to FIG. 7. Numericalvalues shown on the left of FIG. 7 indicate byte numbers, and in thisembodiment, 212 bytes are set to the unit of one frame.

At the position of the zero-th byte positioned at the starting portion,there are recorded block size information of respective bands determinedby the block determining circuits 109, 110, 111 in FIG. 1.

At the position of the first byte, there are recorded information of thenumber of recorded unit blocks or the like. For example, in the bitassignment calculating circuit, it is frequently observed that thehigher band need not be recorded. Concurrently therewith, by decreasingthe number of recorded unit blocks, the bit assignment of the high bandis decreased to zero, and many bits are assigned to middle and low bandswhich are considerably affected from an auditory sense standpoint.Moreover, at the position of the first byte, there are recorded thenumber of unit blocks in which bit assignment information is written ina dual writing fashion and the number of unit blocks in which scalefactor information is written in a dual writing fashion. The dualwriting is a method in which the same data as the data recorded at acertain byte position is recorded on other place for error-correction.Although a strength against error increases in accordance with theincreased number of dual writing information, the number of bits thatcan be used in spectrum data can be decreased. Although the strengthagainst error decreases in accordance with the decreased number of thisdual writing information, the number of bits used in the spectrum datacan be increased. The strength against errors and the number of bitsthat can be used in the spectrum data can be adjusted by setting thenumber of unit blocks in which the dual writing is independentlyeffected on the above-mentioned bit assignment information and the scalefactor information. Incidentally, with respect to each information, thecorrespondence between the code within the determined bit and the numberof the unit block is determined in advance as a format. Specifically, asshown in FIG. 8, for example, of 8 bits at the first byte shown in FIG.7, 3 bits are set to information indicating the number of unit blocksthat are recorded in actual practice, 2 bits of the remaining 5 bits areset to information indicating the number of unit blocks in which bitassignment information is written in a dual writing fashion, and theremaining 3 bits are set to information indicating the number of unitblocks in which the scale factor is written in a dual writing fashion.

At the position of the two bytes in FIG. 7, there is recorded the bitassignment information of the unit block. With respect to the recordingof the bit assignment information, there is determined a format in which4 bits, for example, are used to one unit block. Thus, there arerecorded bit assignment information of the number of unit blocks thatare recorded in actual practice in FIG. 7 in the sequential order of thezero-th unit block.

The scale factor information of unit block is recorded behind the dataof the bit assignment information recorded by the above-mentionedmethod. With respect to the recording of scale factor information, thereis determined a format in which 6 bits, for example, are used to oneunit block. Thus, exactly similarly to the recording of the bitassignment information, there are recorded bit assignment information ofthe number of unit blocks that are recorded in actual practice in thesequential order of the zero-th unit block.

Spectrum data of unit block is recorded behind the scale factorinformation thus recorded. With respect to the spectrum data, there arerecorded spectrum data of the number of unit blocks that are recorded inactual practice in the sequential order of the zero-th unit block. Sincethe number of spectrum data existing at every unit blocks is determinedin advance by the format, it becomes possible to associate data witheach other by the above-mentioned bit assignment information.Incidentally, the unit block in which the bit assignment is zero is notrecorded.

After the above-mentioned spectrum data, the above-mentioned scalefactor is written in a dual writing fashion and the bit assignmentinformation is written in a dual writing fashion. In this recordingmethod, the correspondence of the number is made corresponding to thedual writing information shown in FIG. 12 and others are similar tothose of the scale factor information and the bit assignmentinformation.

In FIG. 7, the scale factor dual writing and/or bit assignmentinformation dual writing may be removed and the resultant extra bits maybe assigned to the spectrum data area.

With respect to the last 2 bytes, as shown in FIG. 7, information ofzero-th byte and first byte are written in a dual writing fashion. Thedual writing of the two bytes is determined as the format, and theamount of the dual writing cannot be changed like the dual writing ofthe scale factor information and the dual writing of the bit assignmentinformation.

That is, in the bit assignment calculation circuit 118 shown in FIG. 1,the system shown in FIG. 3 outputs data in which MDCT output spectrum isprocessed by sub-information as main information and outputs scalefactor indicating state of block floating and word length indicatingword length as sub-information. Based on the above-mentioned data, thescale factor and the word length, the adaptive bit assignment encodingcircuits 106, 107, 108 execute the re-quantization in actual practice,and encode data in accordance with the encoding format.

FIG. 9 shows a decoding circuit for again decoding the high-efficiencycoded signal from the system shown in FIG. 1. Quantized MDCTcoefficients of the respective bands, i.e. data equivalent to the outputsignals of the output terminals 112, 114, 116 shown in FIG. 1 aresupplied to a decoding circuit input terminal 908 in FIG. 9. Block sizeinformation which are information compression parameters used, i.e. dataequivalent to the output signals of the output terminals 113, 115, 117are supplied to the input terminal 910. In the adaptive bit assignmentdecoding circuit 906 shown in FIG. 9, the bit assignment is released byusing adaptive bit assignment information. Then, in the IMDCT (InvertedModified Discrete Cosine Transform) circuits 903, 904, 905 shown in FIG.9, the signal on the frequency axis is transformed into the signal onthe time axis. The time axis signals of these partial bands are decodedinto the whole band signals by band-synthesizing filters, i.e. IQMF(Inverted Quadrature Mirror Filter) circuits 902, 901 which are inverseband-dividing filters.

The normalization information adjustment circuit 911, which will bedescribed below, acts fundamentally similarly to the normalizationinformation adjustment circuit 119 shown in FIG. 1. That is, thenormalization information adjustment circuit 911 is the circuit whichoutputs numerical values for level adjustment by adding and subtractingunit block to normalization information at every two-dimensional block.Also, an adder 909 is an adder for adding numerical values from thenormalization information adjustment circuit 911 to sub-informationindicating the scale factor of the unit block. When the numerical valueoutputted from the normalization information adjustment circuit 911 isnegative, the adder 909 acts as the subtracter. That is, similarly tothe case of the encoding, all the same numerical values from thenormalization information adjustment circuit 911 are added to andsubtracted from normalization information with respect to all unitblocks, thereby making it possible to adjust the level at 2 dB each.With respect to the block that is to be level-adjusted, numerical valuesare outputted from the normalization information adjustment circuit 911at every two-dimensional block and added to and subtracted from thenormalization information, thereby making it possible to independentlyadjust the level at 2 dB each. In the above-mentioned case, the addedand subtracted results are limited in such a manner that the numericalvalues fall within the range of the numerical values of the scale factordetermined by the format. The scale factor value that is level-adjustedby the adder 909 is used in the decoding process executed after theadaptive bit assignment decoding circuit 906 and can be used tolevel-adjust the decoding signal. Also, for example, the scale factorvalue may be read out from a recording medium (not shown), the adjustedscale factor value may be outputted to the terminal 907, and thereby thescale factor value recorded on the recording medium may be re-recordedas the adjusted scale factor value. Information recorded on therecording medium may be changed and re-recorded according to the need.Therefore, by the very simple system, level information recorded on therecording medium can be changed. Also, as described above, the leveladjustment is independently executed at every two-dimensional blocks,thereby making it possible to realize a so-called filter effect.

While both of the encoding circuit and the decoding circuit include thenormalization information adjustment circuits as described above, thepresent invention is not limited thereto and the filter effect can besufficiently demonstrated only the decoding circuit.

Embodiments in which the level is adjusted by using the normalizationinformation adjustment circuit will be described below. For example, byprogressively increasing or decreasing the output value from thenormalization information adjustment circuit, it becomes possible toobtain so-called fade-in and fade-out which are known as the processingof the audio signal. Also, by designating a part of the audio signal,e.g. period in which the recording level is low and cannot be heardwithout difficulty and adding the output value from the normalizationinformation adjustment circuit to the corresponding period, it ispossible to increase only the level of the corresponding period.Conversely, by designating a period in which the recording level is toohigh, it is possible to lower only the level of the correspondingperiod.

Moreover, with respect to an audio signal of a certain piece of music,by analyzing the magnitude of normalization information on the whole,the level adjustment can be carried out and so-called compressor andlimiter effects can be achieved.

Specific examples in which the filter effect can be achieved by usingthe normalization information adjustment circuit will be describedbelow.

For example, the output value from the normalization informationadjustment circuit is set to the block of high band and normalizationinformation of high-band block is decreased, thereby making it possibleto realize the low-pass filter.

Conversely, for example, the output value from the normalizationinformation adjustment circuit is set to the block of low band andnormalization information of low-band block is decreased, thereby makingit possible to realize a high-pass filter.

Similarly, it is clear that the bandpass filter effect and the combfilter effect can be realized within a range of values that the blockcan take.

The output value from the normalization information adjustment circuitis set to the block of the band outside the predetermined band andnormalization information of the block of the band outside thepredetermined band is decreased, thereby making it possible to realize abandpass filter.

Also, by increasing normalization information based on the output valuefrom the normalization information adjustment circuit, it is possible torealize the boost processing within a range of values that the block cantake.

Also, when a digital signal is a digital stereo signal, with respect toindependent digital signals of every channel, normalization informationof all blocks are decreased relative to only one channel, whereby alocalization is changed and a balance processing can be carried out.

Moreover, by progressively changing the frequency on which the filtereffect is achieved, i.e. block using the time as the parameter, it ispossible to realize a wow effect which is a kind of effect processingused in the musical instruments.

Next, a filter will be described with reference to FIGS. 10 and 11 inwhich a low-pass filter is referred to as an example. FIG. 10 shows themanner in which normalization is effected at very unit block. That is,the maximum spectrum data on the frequency axis within the unit block ornormalization nominated information corresponding to MDCT coefficientwhich is the signal component within the two-dimensional blockconcerning the time and the frequency is selected and the selectednormalization nominated number becomes normalization information of thecorresponding unit block. In FIG. 10, normalization information of theunit block in which the block number is 0 becomes 5, normalizationinformation of the unit block in which the block number is 1 becomes 7,normalization information of the unit block in which the block number is2 becomes 9, normalization information of the unit block in which theblock number is 3 becomes 6, and normalization information of the unitblock in which the block number is 4 becomes 4. Since the encoded datahave these normalization information, upon decoding, it is general thatthe decoding is executed based on these normalization information.

On the other hand, FIG. 11 shows the manner in which the maximumspectrum data on the frequency axis within the unit block ornormalization information determined based on the MDCT coefficient areforced to be changed. The normalization information can be forced to bechanged either upon encoding or decoding. An example in whichnormalization information is forced to be changed when encoded datarecorded on the recording medium is decoded will be described. While theencoded data recorded on the recording medium in actual practice isillustrated in FIG. 10, normalization information is forced to be set to0 with respect to the unit blocks in which the block numbers, forexample, are 3 and 4 relative to the encoded data. This can be realizedby adding the negative value to the normalization information in whichthe block numbers are 3 and 4 before the decoding, for example, isexecuted. If the above-mentioned operation is carried out, the unitblocks in which the block numbers are 3 and 4 are decoded under thecondition that they are normalized by the normalization nominated number0. As a consequence, since the unit blocks in which the block numbersare 3 and 4 are decoded based on the normalization nominated number 0 ofthe lowest level, the spectrum data on the frequency axis or the MDCTcoefficient is decoded as the data of low level. If the unit block inwhich the block number is larger contains a higher frequency component,then this operation is equivalent to the case in which the level of thehigh frequency component is cut. That is, normalization information inwhich the unit block numbers are 3 and 4 are forced to be 0, therebymaking it possible to realize a low-pass filter.

In the examples shown in FIGS. 10 and 11, the number of the unit blocksis 5 ranging from 0, 1 to 5, and the number of normalization nominatednumbers is 10 ranging from 0, 1 to 9. According to the format used inthe mini disc, for example, the number of unit blocks is 52 ranging from0, 1 to 51, and the number of normalization nominated numbers is 64ranging from 0, 1 to 64, thereby making it possible to effect a finercontrol. In the above-mentioned case, if normalization information ofthe unit blocks succeeding 20, for example, are set to 0, thereby makingit possible to realize a low-pass filter in which a cutoff frequency isapproximately 5.5 kHz.

Then, embodiments of the digital signal recording apparatus, the digitalsignal reproducing apparatus, the digital signal transmission apparatusand the digital signal reception apparatus including the above-mentionednormalization adjustment circuit will be described with reference toFIGS. 12A, 12B, 13, 14 and FIG. 15.

In FIGS. 12A, 12B, 13, 14 and FIG. 15, reference letter CPU denotes amicrocomputer, and KEY denotes an input key provided in the apparatus oran input key of a remote controller.

Further, an encoder ENC shows a portion which results from removingadders 120, 121, 122 and the normalization adjustment circuit 119 fromthe encoder shown in FIG. 1. Tin denotes an input terminal 100. Adecoder DEC shows a portion which results from removing an adder 909 anda normalization adjustment circuit 911 from the decoder shown in FIG. 9.Tout denotes an output terminal 900.

In FIGS. 12A, 12B and 14, an operation circuit OPE denotes, of theencoder shown in FIG. 1, the adders 120, 121, 122 and the normalizationadjustment circuit 119. In FIGS. 13 and 15, the operation circuit OPEdenotes, of the decoder shown in FIG. 9, the adder 909 and thenormalization adjustment circuit 911.

In the recording apparatus shown in FIG. 12A, an inputted digital signalfrom the input terminal Tin is supplied to the encoder ENC, in which itis encoded, and an encoded output is supplied to the operation circuitOPE.

When a user enters data instructing a function for changing acousticcharacteristics from the input key KEY, the microcomputer CPU controlsthe operation circuit OPE to execute a necessary computation inaccordance with the instruction from the input key KEY. Outputs of theoperation circuit OPE, i.e. outputs 112, 14, 116 of FIG. 1 and theoutputs 113, 115, 117 from the block determining circuit are supplied tothe modulator MOD, in which they are multiplexed and modulated in apredetermined manner or respective outputs are modulated and thenmultiplexed or re-modulated. The modulated signal from the modulator MODis recorded on a recording medium M by a recording head REC comprising amagnetic head and an optical head.

Further, FIG. 12B shows an example of a recording apparatus including ananalyzing circuit ANA for analyzing level of normalization informationfrom the modulator MOD. The analyzing circuit ANA analyzes the level ofthe normalization information, and the level information is transmittedto the microcomputer CPU. The microcomputer CPU controls the operationcircuit OPE for the computation for realizing the compressor or limiterin accordance with the level information and the operation of the userinput key KE.

In the reproducing apparatus shown in FIG. 13, the recorded signal isreproduced by a reproducing head P from the recording medium M shown inFIG. 10. A demodulator DEM demodulates the reproduced signal in responseto the modulation executed by the modulator MOD.

The demodulated outputs from the demodulator DEM, i.e. signalsequivalent to the outputs from the output terminals 112, 114, 116 of theencoder shown in FIG. 1 are supplied to the input of the operationcircuit OPE, i.e. the input terminal 908 of the encoder shown in FIG. 9.At the same time, signals equivalent to the outputs from the outputterminals 113, 115, 117 of the encoder shown in FIG. 1 are supplied tothe input terminal 910 of FIG. 9.

A user enters data instructing the function to change the acousticcharacteristics from the input key KEY. The microcomputer CPU controlsthe operation circuit OPE for executing a necessary computation inaccordance with the instruction from the input means.

The decoder DEC executes decoding in response to the output from theoperation circuit OPE and the signal supplied to the input terminal 910of FIG. 9, and outputs an output digital signal corresponding to theinput digital signal from the input terminal Tin at the output terminalTout.

In the transmission apparatus shown in FIG. 14, the input digital signalfrom the input terminal Tin is supplied to the encoder ENC, in which itis encoded. The output from the encoder ENC is supplied to the operationcircuit OPE. A user enters data instructing the function to change theacoustic characteristics from the input key KEY. The microcomputer CPUcontrols the operation circuit OPE for executing a necessary computationin accordance with the instruction from the input means.

The outputs of the operation circuit, i.e. the outputs 112, 114, 116 ofFIG. 1 and the outputs 113, 115, 117 from the block determining circuitsare supplied to the modulator MOD, in which they are multiplexed andthen modulated in a predetermined manner or the output signals aremodulated and then multiplexed or re-modulated. The modulated signalfrom the modulator MOD is supplied to a transmitter TX, in which it isfrequency-converted and amplified and thereby converted into atransmission signal. The transmission signal is transmitted by atransmission antenna ANT-T which is a part of the transmitter TX.

In the reception apparatus shown in FIG. 15, the transmission signalfrom the transmission antenna ANT-T shown in FIG. 11 is received by areception antenna ANT-R which is a part of the receiver RX. Thereception signal is amplified and inversely frequency-converted by thereceiver RX. The reception signal from the receiver RX is demodulated bythe demodulator DEM in response to the modulation executed by themodulator MOD.

The demodulated outputs from the demodulator DEM, i.e. signalscorresponding to the outputs from the output terminals 112, 114, 116 ofthe encoder of FIG. 1 are supplied to the input of the operation circuitOPE, i.e. the input terminal 908 of the decoder shown in FIG. 9. At thesame time, signals equivalent to the outputs from the output terminals113, 115, 117 of the encoder shown in FIG. 1 are supplied to the inputterminal 910 shown in FIG. 9.

A user enters data instructing the function to change the acousticcharacteristics from the input key KEY. The microcomputer CPU controlsthe operation circuit OPE for executing a necessary computation inaccordance with the instruction from the input means.

The decoder DEC executes decoding in response to the output from theoperation circuit OPE and the signal supplied to the input terminal 910shown in FIG. 9 and outputs an output digital signal corresponding tothe input digital signal from the input terminal Tin at the outputterminal Tout.

Then, an example of a reproducing and recording apparatus will bedescribed with reference to FIG. 16.

In the reproducing and recording apparatus according to the presentinvention, the recorded signal reproduced from the recording medium M bythe playback P is demodulated by the demodulator DEM and data compressedby the ATRA system, for example, corresponding to the encoder outputterminals 112, 114, 116 of FIG. 1 are obtained. The compressed data isinputted to the operation circuit OPE comprising the adder 909 and thenormalization information adjustment circuit 911 of FIG. 9, in which adesired computation is effected on the normalization information andsupplied to the decoder DEC, thereby decoded.

A user enters data instructing the function to change the acousticcharacteristics from the input key KEY. The microcomputer CPU controlsthe operation circuit OPE for executing a necessary computation inaccordance with the instruction from the input means.

Further, in response to the operation of the input key KEY, themicrocomputer CPU stores an address in which the computed compressiondata is recorded in a memory, not shown. The address stored in thememory is stored in an address using pre-grooves in an MD, for example.

In the reproducing and recording apparatus according to the presentinvention, the compression data in which the necessary computation iseffected on the normalization information by the operation circuit OPEis supplied directly to the modulator MOD bypassing the decoder DEC, andrecorded on the recording medium M through a recording head REC. Thatis, through a short loop comprising the recording medium M, thereproducing head P, the demodulator DEM, the operation circuit OPE, themodulator MOD, the recording head REC and the recording medium M, inthat order, the compression decoding/encoding is not executed but adesired processing is executed on the recording signal on the recordingmedium, and the overwriting is carried out using the address stored inthe memory as a starting point.

Further, according to the present invention, since the processed contentis recorded on then recording medium M, even when the recording mediumis reproduced by other reproducing equipment, there can be obtained datahaving changed acoustic characteristics.

The present invention is not limited to the above-mentioned embodimentsand various modifications and variations can be made. The encoder andthe decoder may be either separately provided or may be integrallyformed. The recording apparatus and the reproducing apparatus may beeither separately provided or may be integrally formed. As the recordingmedium, there may be used a magnetic tape, a magnetic disk, a magnetooptical disk or the like. Further, the recording medium may be replacedwith memory means such as IC memory and memory card. A transmission linebetween the transmission apparatus and the reception apparatus may beestablished by light such as infrared rays or radio transmission linesuch as radio waves or wire transmission line such as conductors andoptical cables. The input digital signal is the digital audio signal,and as the audio signal, there may be used a variety of sound signalssuch as voices, songs and sounds of musical instruments. Further, thedigital audio signal also is possible. The present invention can beapplied to the digital signal recording and reproducing method orapparatus, the digital signal transmission and reception method orapparatus and the digital signal reception method or apparatus or thelike.

In the specific example according to the present invention, with respectto the level adjustment effect or the filter effect achieved by theadjustment of the normalization information, there have been describedthe methods of WMF band-limiting and the coding system using thequadrature transform based on the MDCT. However, the present inventionis not limited to the encoding system using the QMF and the MDCTprocessing. Fundamentally, as long as the encoding system is of thesystem in which the quantization is similarly carried out bynormalization information and bit assignment information, even in thecase of the subband coding using a filter band or the like, for example,it is apparent that the level adjustment effect or the filter effect canbe realized by using a similar method.

According to the present invention, it is needless to say that thevolume processing and the filter processing are effected on thenreproduced digital signal upon reproduction, Upon encoding, if theabove-mentioned processing is effected on the digital signal, then itbecomes possible to record the digital signal on the recording mediumunder the condition that a desired effect such as the volume processingor the filter processing is reflected on the digital signal. That is,not only music information from other sound source can be processed in adesired processing fashion and recorded on the recording medium but alsodata reproduced from the recording medium can be processed in a desiredprocessing fashion and then re-recorded on the recording medium.

Further, according to the present invention, since the normalizationinformation of every unit block is added or subtracted prior to theATRAC decoding processing, it becomes possible to execute the volumeprocessing or the filter processing by adding the adder or the adderwhich becomes the subtracter when the positive value or the negativevalue is added.

For example, when spectrum data is directly computed after compressiondata was restored to spectrum data, inputted digital data on the timeaxis is processed in an MDCT fashion and computed and then compressed.Therefore, it is necessary to add a circuit for computing the spectrumdata during the processing of the ATRAC decoder, and hence the ATRACdecoder should be modified. According to the present invention, sincedata is decoded by the ATRAC decoder and the addition or the subtractionis effected on the normalization information, the ATRAC decoder may bethe original one and need not be modified as described above.

In the case of the encoding, similarly, after the encode processing, thevolume processing or the filter processing can be made by thecomputation based on the adder. Thus, the addition of the circuit foreffecting the computation during the processing of the ATRAC encoder isnot required, and hence the ATRAC encoder need not be modified.

A reproducing and recording method will be described with reference to aflowchart of FIG. 17.

Referring to FIG. 17, following the start of operation, control goes toa step 11, whereat compressed digital data including spectrum data,band-divided on the frequency axis, and scale factor of every dividedband is read out from the recording medium. Then, control goes to thenext decision step 12, whereat it is determined whether or not thechange of acoustic characteristics is instructed by a user when the useroperates the key. If the change of acoustic characteristics isinstructed as represented by a YES at the decision step 12, then controlgoes to a step 13, whereat a starting address of data whose acousticcharacteristics are to be changed is stored in the memory. Control goesto a step 14, whereat a computation for changing the acousticcharacteristics of the compressed digital data is effected on the scalefactor of each divided band of the compressed digital data having theband-divided spectrum data on the frequency axis in which the change ofthe acoustic characteristics was instructed and the scale factor ofevery divided band. At a step 15, the recording head comprising themagnetic head and the optical head, for example, is moved based on thestarting address stored at the step 13. At a step 16, by effecting acomputation on the scale factor of every band, the compressed digitaldata whose acoustic characteristics are changed is overwritten from thestarting address. If the change of the acoustic characteristics is notinstructed as represented by a NO at the decision step 12, then controlgoes to a step 17, wherein compressed digital data, which is notcomputed, can be recorded one more time.

A decoding method will be described with reference to a flowchart ofFIG. 18.

Referring to FIG. 18, and following the start of operation, it isdetermined at the next decision step 21 whether or not the change ofacoustic characteristics is instructed by a user when the user operatesthe key. If the change of acoustic characteristics is instructed asrepresented by a YES at the decision step 12, then control goes to astep 22. If the change of acoustic characteristics is not instructed asrepresented by a NO at the decision step 21, then control goes toe astep 23. In the step 22, a predetermined computation is effected on thescale factor of each divided band of the compressed digital data havingthe band-divided spectrum data on the frequency axis in which the changeof the acoustic characteristics was instructed and the scale factor ofevery divided band. In the step 23, based on the scale factor of everydivided band thus computed, there is normalized the band-dividedspectrum data in the compressed digital data. In the step 24, digitaldata on the band-divided time axis is obtained by processing theband-divided spectrum data thus normalized in an IMDCT fashion. In thenext step 25, band-divided digital data on the time axis are synthesizedin band.

A recording method will be described with reference to a flowchart ofFIG. 19.

Referring to FIG. 19, and following the start of operation, at a step31, a DCT is effected to transform the inputted digital signal on thetime axis into the spectrum data on the frequency axis. At a step 32,the spectrum data on the frequency axis is band-divided into a pluralityof bands, and a scale factor of every divided band used fornormalization is computed. At a step 33, in response to the calculatedscale factor of every divided band, the spectrum data thus band-dividedon the frequency axis is compressed and thereby transformed intocompressed digital data including the scale factor of every divided bandand the spectrum data. At the next decision step 34, it is determinedwhether or no the change of acoustic characteristics is instructed whena user operates the input key. If the change of acoustic characteristicsis instructed as represented by a YES at the decision step 34, thencontrol goes to a step 35. If on the other hand the change of acousticcharacteristics is not instructed as represented by a NO at the decisionstep 34, then control goes to a step 37. In the step 35, the compresseddigital data including the scale factor of every divided band and thespectrum data is inputted and a computation for changing the acousticcharacteristics of the compressed digital data is effected on the scalefactor of every divided band. In a step 36, the compressed digital datain which the acoustic characteristics are changed by the computation onthe scale factor of every band is recorded on the recording medium. Ifthe change of acoustic characteristics is not changed as represented bya NO at the decision step 34, then control goes to the step 37, whereatcompressed digital data, which is not computed, is recorded.

Having described preferred embodiments of the invention with referenceto the accompanying drawings, it is to be understood that the inventionis not limited to those precise embodiments and that various changes andmodifications could be effected therein by one skilled in the artwithout departing from the spirit or scope of the invention as definedin the appended claims.

What is claimed is:
 1. A reproducing and recording apparatus comprising:data read means for reading compressed digital data including spectrumdata whose band is divided into a plurality of bands on a frequency axisand a scale factor of every divided band from a recording medium;computation means for receiving compressed digital data including saidband-divided spectrum data and the scale factor of every divided bandfrom said data read means and effecting a computation for changingacoustic characteristics of said compressed digital data; and recordingmeans for re-recording said compressed digital data whose acousticcharacteristics are changed when said computation means computes thescale factor of said every band on said recording medium.
 2. Theapparatus of claim 1, further comprising: normalization means fornormalizing said band-divided spectrum data contained in said compresseddigital data based on the scale factor of said every divided bandreceived by said computation means; IMDCT means for obtainingband-divided digital data on a time axis by processing said band-dividedspectrum data normalized by said normalization means in an IMDCTfashion; and band-synthesizing means for band-synthesizing said digitaldata on the time axis band-divided by said IMDCT means.
 3. The apparatusas claimed in claim 2, wherein said computation means executes apredetermined computation to change acoustic characteristics of saiddigital data on the time axis band-synthesized by said band-synthesizingmeans.
 4. The apparatus as claimed in claim 2, wherein digital data on atime axis band-synthesized by said band synthesizing means is a digitalaudio signal comprising a plurality of independent channels and saidcomputation means changes a localization of the digital audio signalcomprising said plurality of independent channels by effecting acomputation on at least one channel of said independent channels.
 5. Theapparatus as claimed in 1, wherein said computation means effectsuniformly a computation on all of said scale factors of said everydivided band.
 6. The apparatus as claimed in claim 1, wherein saidcomputation means effects a computation on at least one of said scalefactors of said every divided band.
 7. The apparatus as claimed in claim1, wherein said computation means executes a filter processing byrelatively changing a scale factor corresponding to one band of saidplurality of bands and a scale factor corresponding to other band. 8.The apparatus as claimed in claim 7, wherein one of said plurality ofbands has a frequency band higher than that of other band and saidcomputation means executes a low-pass processing by effecting acomputation such that a scale factor corresponding to one band of saidplurality of bands decreases relative to a scale factor corresponding toother band.
 9. The apparatus as claimed in claim 7, wherein one band ofsaid plurality of bands has a frequency band higher than that of otherband and said computation means executes a high-pass processing byeffecting a computation such that a scale factor corresponding to oneband of said plurality of bands increases relative to a scale factorcorresponding to other band.
 10. The apparatus as claimed in claim 7,wherein said computation means executes a bandpass filter processing byeffecting a computation such that a scale factor corresponding to oneband of said plurality of bands increases relative to scale factorscorresponding to other band adjoining to both sides of one band.
 11. Theapparatus of claim 1, wherein said compressed digital data isvolume-controlled by uniformly effecting a computation on all scalefactors of said every divided band.
 12. The apparatus as claimed inclaim 1, wherein said compressed digital data is filter-processed byeffecting a computation on at least one of scale factors of said everydivided band.
 13. The apparatus as claimed in claim 1, whereincompressed digital data of a predetermined period is relatively changedwith respect to compressed digital data of other period and recorded byeffecting a computation for changing the acoustic characteristics ofsaid compressed digital data relative to said compressed digital data ofsaid predetermined period.
 14. The apparatus according to claim 1,further comprising level analyzing means for analyzing level informationof said compressed digital data and wherein said computation meanseffects a computation for changing said acoustic characteristics inaccordance with analyzed results of said level analyzing means tothereby limit the level of said compressed digital data.
 15. Areproducing and recording method comprising the steps of: readingcompressed digital data including spectrum data on a frequency axisband-divided and a scale factor of every divided band from a recordingmedium; effecting a computation for changing acoustic characteristics ofsaid compressed digital data on the scale factor of said every dividedband in compressed digital data including said read out spectrum data onsaid frequency axis band-divided and the scale factor of every dividedband; and re-recording said compressed digital data whose acousticcharacteristics are changed on said recording medium by computing thescale factor of said every band.
 16. The method of claim 15, furthercomprising the steps of: normalizing said band-divided spectrum datacontained in said compressed digital data based on the scale factor ofsaid every divided band of said read out compressed digital data;obtaining band-divided digital data on a time axis by processing saidnormalized band-divided spectrum data; and band-synthesizing saiddigital data on the band-divided time axis.
 17. The method as claimed inclaim 16, wherein said computation is a predetermined computation tochange acoustic characteristics of said digital data on theband-synthesized time axis.
 18. The method as claimed in claim 16,wherein digital data on a band-synthesized time axis is a digital audiosignal comprising a plurality of independent channels and saidcomputation changes a localization of the digital audio signalcomprising said plurality of independent channels by effecting acomputation on at least one channel of said independent channels. 19.The method as claimed in 15, wherein said computation is a uniformcomputation on all of said scale factors of said every divided band. 20.The method as claimed in claim 15, wherein said computation is acomputation on at least one of said scale factors of said every dividedband.
 21. The method as claimed in claim 15, wherein said computation isa filter processing by relatively changing a scale factor correspondingto one band of said plurality of bands and a scale factor correspondingto other band.
 22. The method as claimed in claim 21, wherein one ofsaid plurality of bands has a frequency band higher than that of otherband and said computation is a low-pass processing by effecting acomputation such that a scale factor corresponding to one band of saidplurality of bands decreases relative to a scale factor corresponding toother band.
 23. The method as claimed in claim 21, wherein one band ofsaid plurality of bands has a frequency band higher than that of otherband and said computation is a high-pass processing by effecting acomputation such that a scale factor corresponding to one band of saidplurality of bands increases relative to a scale factor corresponding toother band.
 24. The method as claimed in claim 21, wherein saidcomputation is bandpass filter processing by effecting a computationsuch that a scale factor corresponding to one band of said plurality ofbands increases relative to scale factors corresponding to other bandadjoining to both sides of one band.
 25. The method of claim 15, whereinsaid compressed digital data is volume-controlled by uniformly effectinga computation on all scale factors of said every divided band.
 26. Themethod as claimed in claim 15, wherein said compressed digital data isfilter-processed by effecting a computation on at least one of scalefactors of said every divided band.
 27. The method as claimed in claim15, wherein compressed digital data of a predetermined period isrelatively changed with respect to compressed digital data of otherperiod and recorded by effecting a computation for changing the acousticcharacteristics of said compressed digital data relative to saidcompressed digital data of said predetermined period.
 28. The methodaccording to claim 15, further comprising the step of analyzing levelinformation of said compressed digital data; and wherein saidcomputation changes said acoustic characteristics in accordance with theanalyzed level information to thereby limit the level of said compresseddigital data.